1. Field of the Invention
The present invention relates to receivers and in particular to synchronizing many receivers to a master clock.
2. Description of the Related Art
There is an increasing demand for new technologies and innovative projects in the field of entertainment electronics. It is an important precondition for the success of new multimedia systems to offer optimum functionalities or capabilities, respectively. This is achieved by the use of digital technologies and in particular computer technology. Examples for this are the applications offering an improved audio-visual impression close to reality. In present audio systems a main drawback is the quality of spatial acoustic reproduction of natural but also virtual environments.
Methods for a multi-channel loudspeaker reproduction of audio signals have been known and standardized for many years. All conventional technologies have the disadvantage that both the set-up location of the loudspeaker and the position of the listener are already embossed onto the transmission format. In a wrong arrangement of the loudspeakers with regard to the listener, audio quality clearly suffers. An optimum sound is only possible in a small range of the reproduction domain or space, the so-called sweet spot.
small range of the reproduction domain or space, the so-called sweet spot.
A better natural domain impression and a stronger enveloping in audio reproduction may be achieved with the help of a new technology. The bases of this technology, the so-called wave-field synthesis (WFS) were researched at the TU Delft and presented first in the late 80ies (Berkhout, A. J.; de Vries, D.; Vogel, P.: Acoustic control by Wave-field Synthesis. JASA 93, 1993).
As a consequence of the enormous requirements of this method with regard to computer performance and transmission rates, the wave-field synthesis was hitherto only rarely used in practice. Only the advances in the fields of microprocessor technology and audio encoding allow today the use of this technology in concrete applications. First products in the professional field are expected next year. In a few years, also first wave-field synthesis applications for the consumer area are to enter the market.
The basic idea of WFS is based on the application of the Huygens' principles of wave theory:
Every point on a propagating wavefront serves as the source of a wavelet propagating in a spherical or circular form, respectively.
If applied to acoustics, by a large number of loudspeakers arranged next to each other (a so-called loudspeaker array) any form of an incoming wavefront may be reproduced. In the simplest case, an individual punctual source to be reproduced and a linear arrangement of the loudspeakers, the audio signals of every loudspeaker have to be supplied with a time delay and an amplitude scaling so that the reflected sound fields of the individual loudspeakers are correctly overlaid. With several acoustic sources, for each source the contribution for each loudspeaker is calculated separately and the resulting signals are added. If the sources to be reproduced are in a room with reflecting walls, then also reflections have to be reproduced as additional sources via the loudspeaker array. The effort in calculating thus strongly depends on the number of acoustic sources, on the reflection characteristics of the recording room, and on the number of loudspeakers.
The advantage of this technology is in particular that a natural spatial sound impression is possible via a large area of the reproduction domain. In contrast to known technologies, direction and distance of acoustic sources are reproduced very accurately. In a limited way, virtual acoustic sources may even be positioned between the real loudspeaker array and the listener.
In the practical implementation problems result insofar that after the audio signals have been calculated for the individual loudspeakers, the same are distributed to the individual loudspeakers and then synchronously reproduced by the individual loudspeakers. As it was explained above, the individual loudspeakers have to be fed so that the signals output by the same correctly overlay in order to reconstruct an original “large” wave by overlaying many “small” waves so that a listener thinks that the “large” wave comes from an acoustic source arranged in another position and not from many individual loudspeakers, which respectively output a “small” wave. For this purpose it is of a decisive importance that the individual loudspeakers operate synchronously so that the individual wave calculated by the wave-field synthesis means are reproduced correctly, i.e. correctly converted into acoustic waves.
Typical systems operate digitally so that one sequence of digital samples is respectively supplied to the individual loudspeakers. A mutual synchronity of the individual loudspeakers is achieved when all loudspeakers are operated using the same sampling clock or “resampling” clock.
At this point it is to be noted that the problematic of synchronization also exists in many other places of audio technology. In live transmissions there is further the requirement that a sampling clock in the transmitter recording the audio scene to be transmitted is synchronous to the sampling clock in the receiver reproducing the transmitted audio scene. If the recording sampling clock and the reproduction sampling clock do not operate synchronously, samples would accumulate somewhere on the transmission path if the reproduction clock is too slow, or would run out if the reproduction clock is too fast. In order to mitigate this situation, buffers are built in, so that a certain deviation corresponding to the buffer size is allowed between the recording clock and the reproduction clock. Such buffers are usually built into the receiver, wherein the reproduction in the receiver takes place somewhat offset in time such that a reproduction only starts with a receiver buffer filled up to a certain degree. In such a case the recording clock and the reproduction clock may vary in certain limits determined by buffer size.
If such a concept would be transferred to the synchronization of many receivers in a wave-field synthesis, this would lead to a loss of audio quality, as no samples run out or no samples accumulate during transmission. There is, however, no control as to whether all receivers operate synchronously, i.e. that all receivers output their samples to their corresponding loudspeaker exactly at the point of time predetermined by the wave-field synthesis means.
In recent times, the so-called firewire data transmission format was proposed for a real-time transmission of audio signals. For this, reference is made to the expert's publication “Sample clock jitter and real-time audio over the IEEE 1394 high performance serial bus”, Julian Dunn, 106. AES-Convention, May 8 to 11, 1999, Munich, Preprint 4920. In this expert's publication it is noted that in the IEEE 1394 data stream a sample jitter of 40 ns is to be expected and that further a jitter frequency occurs depending on the frequency deviation between usually the crystal oscillator in the transmitter and the crystal oscillator in the receiver. Usually, each firewire node, i.e. each transmitter or receiver, respectively, includes a free-running crystal oscillator having 24.576 MHz. This clock is used in order to increment a cycle-time register in each node. The node defined as cycle master transmits a cycle start packet in intervals of 125 μs, i.e. with a frequency of 8 kHz. This start packet defines the start of an isochronous cycle according to IEEE 1394. This packet has a value enabling other nodes on the bus to align their cycle-time registers in order to correct a drifting due to somewhat different clock frequencies. After a cycle start packet is transmitted on the bus, it is subjected to a reclocking jitter, so that in the cycle-time register alignment also a jitter results.
Based on these jitter problems and the temporarily varying jitter frequency, during the audio reproduction in multi-channel audio systems and in particular in wave-field synthesis applications audible artefacts occur. If, alternatively, a slow PLL is used in order to attenuate the jitter in the audio reproduction clock, the firm delay between input and output is lost, which may lead to a sample de-synchronization of the individual receivers and in the worst case even to a loss of samples. In multi-channel applications, therefore a fixed timing, i.e. a fixed synchronization between individual channels, is a substantial requirement.